麥克風陣列對移動音源之相對角度追蹤演算法設計

以作者查詢圖書館館藏

、以作者查詢臺灣博碩士

、以作者查詢全國書目

、勘誤回報

、線上人數：45

、訪客IP：3.134.118.95

姓名

張慶霄(Ching Hsiao Chang) 查詢紙本館藏

畢業系所

通訊工程學系在職專班

論文名稱

麥克風陣列對移動音源之相對角度追蹤演算法設計
(An Algorithm Design for the Microphone Array to Track the Relative Arrival Angle of a Mobile Audio Source)

相關論文

★ 利用手持式手機工具優化行動網路系統於特殊型活動環境	★ 穿戴裝置動態軌跡曲線演算法設計
★ 石英諧振器之電極面設計對振盪頻率擾動之溫度相依性研究	★ 股票開盤價漲跌預測
★ 感知無線電異質網路下以不完美頻譜偵測進行資源配置之探討	★ 大數量且有限天線之多輸入多輸出系統效能分析
★ 具有元學習分類權重轉移網路生成遮罩於少樣本圖像分割技術	★ 具有注意力機制之隱式表示於影像重建三維人體模型
★ 使用對抗式圖形神經網路之物件偵測張榮	★ 基於弱監督式學習可變形模型之三維人臉重建
★ 以非監督式表徵分離學習之邊緣運算裝置低延遲樂曲中人聲轉換架構	★ 基於序列至序列模型之 FMCW雷達估計人體姿勢
★ 基於多層次注意力機制之單目相機語意場景補全技術	★ 應用於3GPP WCDMA-FDD上傳鏈路系統的遞迴最小平方波束合成犛耙式接收機
★ 調適性遠時程瑞雷衰退通道預測演算法設計與性能比較	★ 智慧型天線之複合式到達方位-時間延遲估測演算法及Geo-location應用

檔案

[Endnote RIS 格式]

[Bibtex 格式]

[相關文章]

[文章引用]

[完整記錄]

[館藏目錄]

[檢視]

[下載]

本電子論文使用權限為同意立即開放。
已達開放權限電子全文僅授權使用者為學術研究之目的，進行個人非營利性質之檢索、閱讀、列印。
請遵守中華民國著作權法之相關規定，切勿任意重製、散佈、改作、轉貼、播送，以免觸法。

摘要(中)

本篇論文運用音源和麥克風陣列間，快速的相對運動，所造成的訊號到達時間差之補償，來解決移動中的聲音訊號來源的追蹤問題。並藉由模擬不同的麥克風陣列排列形狀來比較追蹤角度的精確度。
其中利用sinc函數所構成的位移、取樣、延遲濾波器，及麥克風陣列和音源，在笛卡爾座標系中的相對座標，經由X軸及Y軸的旋轉及疊代最小平方誤差演算法(LMS)，來補償麥克風陣列間各個麥克風的接收訊號延遲誤差，進而解決移動中的聲音訊號來源的追蹤問題。
而sinc函數延遲濾波器的校正值，是運用調適性訊號處理(ASP)中的疊代最小平方誤差演算法(LMS)，來獲得麥克風陣列間各個麥克風的修正角度的最小誤差標準，再得到每次疊代的更新補償角度。此補償角度的獲得，是得自於相對應的麥克風陣列在三維的笛卡兒座標系中的旋轉追蹤角度。而此旋轉追蹤角度，所指的是，假設此麥克風陣列在笛卡兒座標系中動態的旋轉調整，用以追蹤快速運動的音源，使其訊號波以零相角差，到達此麥克風陣列的座標原點，所需的旋轉角度。相對於初始的麥克風陣列位置，此旋轉追蹤角度，既等於訊號到達角度(DOA)的估測值。也就是聲音訊號來源到達麥克風陣列的座標原點的二維相對方位。再藉由不同的麥克風陣列排列形狀來研究追蹤角度的精確度改善。

摘要(英)

This thesis investigates the use of microphone arrays to solve the tracing problem of the moving audio source with the rapid relative movement caused by the signal arrival time difference. The simulation results are presented to demonstrate the tracking angle accuracy with different array configurations.
The sinc functions are utilized to construct the shifting, sampling, and delay filter. In the Cartesian coordinate system with the relative coordinates, through the X-axis and Y-axis rotation, the iterative least mean squares error algorithm (LMS) is employed to compensate for microphone array receiving signals with the various microphone delay errors such that it solves the tracing of the rapid moving audio source.
The compensation of the sinc function filter is to use the adaptive signal processing of the iterative least mean squares error algorithm (LMS) to get the minimum compensation angle for each microphone array element, and then the scheme obtains the compensation point of each iteration in the update. The compensation for the acquisition of this point of view is derived from the corresponding three-dimensional Cartesian coordinate system to track the angle of the rotation in the microphone array. Regarding the rotation tracking angle, it is achieved by adjusting the Cartesian coordinate to make no phase difference. Relative to the initial position of the microphone array, this rotation tracking angle is equal to the estimated direction of signal arrival (DOA) estimation of the value. By various micro array configurations, the improvement on the accuracy is investigated.

關鍵字(中)

★ 麥克風陣列
★ 音源追蹤
★ 最小平方誤差演算法
★ 訊號到達角度

關鍵字(英)

★ rapid moving audio source
★ LMS
★ microphone array
★ audio tracking angle

論文目次

摘要 i
Abstract iii
誌謝 v
目錄 vi
圖目錄 viii
表格目錄 x
第一章緒論 1
1.1 研究動機 1
1.2 研究目的 3
1.3 論文架構 4
第二章利用訊號到達麥克風陣列時間差(TDOA)追蹤快速移動聲音訊號源的演算法 7
2.1 由sinc函數構成的調適性訊號延遲濾波器 9
2.2 移動音源和麥克風陣列之間的訊號到達時間差的最小平方誤差疊代補償演算法 13
2.2.1 表示麥克風陣列移動的笛卡兒座標系 14
2.2.2 訊號到達角度最小均方誤差疊代(RLMS)補償演算法 16
第三章化簡“利用訊號到達麥克風陣列時間差(TDOA)的互動補償方法 28
3.1 用查表法簡化函數的微分的運算量 28
3.2 簡化笛卡爾座標系的轉換矩陣運算量 32
第四章以MATLAB模擬“利用訊號到達麥克風陣列時間差(TDOA)追蹤快速移動聲音訊號源的方法” 34
4.1 以MATLAB 模擬演算法的流程圖: 34
4.2 模擬步驟: 36
第五章比較不同排列形狀的麥克風陣列模擬結果 42
第六章延伸應用 54
6.1 三維排列的麥克風陣列來擴大訊號來源方向偵測範圍及訊號來源角度估測的精度 54
6.2 用兩個陣列估測出的訊號到達角度及訊號源距離 56
6.3 多媒體3D音效模擬 57
參考文獻 58

參考文獻

[1] Y.T. Chan and K.C. Ho, “TDOA-SDOA estimation with moving source and receivers,” Proc. IEEE ICASSP., vol.42, no.8, pp.1905-1915, Aug. 1994.
[2] Y.T. Chan and K.C. Ho, “A simple and efficient estimator for hyperbolic location,” IEEE Trans. Signal Process., vol.42, no8, pp.1905-1915, Aug. 1994.
[3] M. Zhang and M.H. Er, ‘An alternative algorithm for estimating and tracking talker location by microphone arrays,”J. Audio Eng. Soc., vol.44, no.9,pp.729-736, Sept. 1996.
[4] S. Affes, S. Gazor, and Y. Grenier, “Robust adaptive beamforming via LMS-like target tracking,” Proc. IEEE ICASSP, vol.4, pp.269-272, 1994.
[5] D.M. Etter and S.D. Stearns, “Adaptive estimation of time delays in sampled data systems,” IEEE Trans. Acoust. Speech Signal Process., vol.29, no.3, pp.582-587, June 1981.
[6] P.C. Ching and Y.T. Chan, “Adaptive time delay estimation with constrains,” IEEE Trans. Acoust. Speech Signal Process., vol.36, no.4, pp.599-602, April 1988.
[7] H.C. So, P.C. Ching, and Y.T. Chan, “A new algorithm for explicit adaptation of time delay,” IEEE Trans. Signal Process., vol.42, no.7, pp.1816-1820, July 1994.
[8] P.L. Feintuch, N.J. Bershad, and F.A. Reed, “Time delay estimation using the LMS adaptive filter-Dynamic behavior,” IEEE Trans. Acoust. Speech Signal Process., vpl.29, no.3, pp.571-576, June 1981.
[9] Y.T. Chan, J. Riley, and J.B. Plant, “Modeling of time delay and its application to estimation of non-stationary delays,” IEEE Trans. Acoust. Speech Signal Process., vol.29, no.3, pp.577-581, June 1981.
[10] Y.T. Chan, J. Riley, and J.B. Plant, “A parameter estimation approach to time delay estimation and signal detection,” IEEE Trans. Acoust. Speech Signal Process., vol.28, no2, pp.8-16, Feb. 1980
[11] D. M. Etter and S. D. Steams, “Adaptive estimation of time delays in sampled data systems,’’ IEEE Trans. Acousr., Speech, Signal Processing, vol. ASSP-29, pp. 582-587, June 1981.
[12] “Special issue on time delay estimation,” IEEE Trans. Acoust., Speech, Signal Processing, vol. ASSP-29, June 1981.
[13] B. Widrow and S. D. Steams, Adaptive Signal Processing. Englewood Cliffs, NJ: Prentice-Hall, 1985.
[14] Toshihnru Horiuchi, Mitsunori Mizumnchi, and Satoshi Nakamum, “ITERATIVE COMPENSATION OF MICROPHONE ARRAY AND SOUND SOURCE OVEMENTS BASED ON MINIMIZATION OF ARRIVAL TIME DIFFERENCES” 2004 lEEE Sensor Array and Multichannel Signal Processing Workshop.
[15] C. H. Knapp and G. C. Carter, “The generalized correlation method for estimation of time delay,” IEEE Trans. Acoust., Speech & Signal Process., vol. 24, no. 4, pp. 320-327, Aug. 1976.
[16] M. Omologo and P. Svaizer, “Use of the Crosspower-Spectrum Phase in Acoustic Event Localization,” lEEE Trans. Speech & Audio Process., vol. 5, no. 3, pp. 288-292, May 1997.
[17] H. Kagiwada, H. Ohmori, and A. Sano, “A recursive algorithmfor tracking DOA’s of multiple moving targets by using linear approximations,” IEICE Trans. Fundamentals, vol. E81-A, no. 4, pp. 639-648, Apr. 1998.
[18] Y. T. Chan and K. C. Ho, “A simple and efficient estimator for hyperbolic location,” IEEE Trans. Signal Process., vol. 42, no. 8, pp. 1905-1915, Aug. 1994.
[19] M. S. Brandstein, J. E. Adcock, and H. F. Silverman, “A closed-form location estimator for use with roam environment microphone arrays,” IEEE Trans. Speech & Audio Process., vol. 5, no. 1, pp. 45-50, Jan. 1997.
[20] Percival, Donald B. and Andrew T. Walden. Spectral Analysis for Physical Applications. Cambridge University Press, 1993.
[21] Pandit, Sudhakar M. and Wu, Shien-Ming. Time Series and System Analysis with Applications. John Wiley & Sons, Inc., 1983.
[22] G. Udny Yule On a Method of Investigating Periodicities in Disturbed Series, with Special Reference to Wolfer's Sunspot Numbers Philosophical Transactions of the Royal Society of London, Ser. A, Vol. 226, (1927) 267--298.
[23] Gilbert Walker On Periodicity in Series of Related Terms, Proceedings of the Royal Society of London, Ser. A, Vol. 131, (1931) 518--532.
[24] Toshihnru Horiuchi, Mitsunori Mizumnchi, and Satoshi Nakamum ITERATIVE COMPENSATION OF MICROPHONE ARRAY AND SOUND SOURCE MOVEMENTS BASED ON MINIMIZATION OF ARRIVAL TIME DIFFERENCES lEEE Sensor Array and Multichannel Signal Processing Workshop 2004
[25] Y. T. Chan, J. M. F. Riley, and J. B. Plant, “Modeling of time delay and its application to estimation of nonstationary delays,’’ IEEE Trans. Acousr., Speech, Signal Processing, vol. ASSP-29, pp. 577-581, June 1981
[26] H. C. So and P. C. Ching, “A novel constrained time delay estimator,” in Proc. 1993 Inr. Conf. Signal Processing (Beijing, China), Oct. 26-30, 1993.
[27] J. R. Treichler, “Transient and convergent behaviour of the adaptive line enhancer,” IEEE Trans. Acoust. Speech Signal Processing, vol. ASSP-27, pp. 53-62, Feb. 1979.

指導教授

陳永芳(Yung-Fang Chen)

審核日期

2010-7-6

推文